The present invention relates to an information coding method utilizing focused error correction and/or error detection, in which method the quality of the data transfer connection is used for selecting the coding mode for the data transfer connection. The invention also relates to a system and terminal devices applying the method. The invention is particularly suitable for use in connection with data transfer connections realized by radio.
While transferring information, such as speech or data, using transfer connections subject to transmission errors, the information to be transferred is in general protected using an error correction algorithm. Especially in digital connections an attempt is made to detect transmission errors, and to correct the erroneous information bits. How successfully this is done depends among other things on the number of transmission errors and on the error correction algorithm used. In speech coding systems prior known to a person skilled in the art, a major part of the bits comprising speech information are protected using an error correction code. This is the procedure e.g. in the so called full rate (FR, Full Rate) speech codec of the GSM system.
In the full-rate speech codec (which later is also called FR-speech codec) of the GSM system a RPE-LTP (Regular Pulse Excitation-Long term Prediction) based speech encoding system is used. It produces 260 speech parameter bits for each 20 ms speech frame. Out of these 260 bits, the 182 subjectively most important bits are protected using an error correction code. As the error correction code, xc2xd-rate convolution encoding is used. The remaining 78 bits are transferred in the data transmission connection completely without error correction.
The number of transmission errors on a data transfer connection may temporarily exceed the error correction capacity of the xc2xd-rate convolution coding used in the GSM-system. As a result, the important received speech parameter bits may contain transmission errors. It is important to detect that these transmission errors occurred, even if it were not possible to correct them. If the speech parameters which are the most important for speech quality contain transmission errors, they shall not be used for speech synthesizing in the receiver, but they must be rejected. In the full rate FR-speech codec of the GSM-system 3-bit CRC (Cyclic Redundancy Check)-error detection is used. CRC-error detection is focused on the 50 most important bits of speech coding. In a receiver the error detection code is used for verifying the correctness of the 50 most important bits of each 20 ms speech frame. If they contain errors, the frame is classified as bad and it is not used in speech synthesizing. Instead, an attempt is made to substitute the bad frame with an estimate, which is formed e.g. based upon chronologically preceding error-free frames.
The full-rate speech coding method of the GSM-system briefly presented above, operates reasonably well, provided that the relative share of transmission errors does not grow too high. Under these conditions the error correction algorithm is capable of correcting transmission errors sufficiently for obtaining a satisfactory transfer connection and through it a satisfactory speech quality. When the proportion of transmission errors grows to medium or high level, the error correcting capability of convolution coding having xc2xd-rate coding ratio is exceeded. In this case a more efficient error correction algorithm would be needed, such as e.g. a convolution coding having ⅓-rate coding ratio. In this case, however, the total speech encoding efficiency will be reduced essentially, because more error correction information bits must be included in the data transfer connection. This naturally increases the data transfer rate required of the data transfer connection. Accordingly, this approach cannot be used for codecs with fixed line speed. Instead, the above presented method based upon making the error correction algorithm more efficient is suitable for systems with variable line speed.
For instance, the total bit rate of the data transfer system used for transferring speech can be kept constant, provided that at the same time when the number of bits used for the error correction of speech parameter bits is increased, the number of bits used for speech encoding itself is reduced. This in turn requires using several different speech codecs with different line speeds in both the transmitter and the receiver, which makes the structure of the system more complicated. Further, the lower the number of bits used for speech encoding, the more calculation capacity is normally required of the various components of the system. The above presented disadvantages increase the cost of the system. In addition to the above, the deterioration of speech quality cannot be avoided when more bits are used for error correction, because the fewer bits there are available for speech encoding, the more one has to compromise the voice quality. The voice quality deterioration due to the reduction of the number of bits used for speech encoding is particularly important in a case where there is background noise to speech, e.g. the noise from a car engine.
One problem occurring in speech coding methods according to prior art is the complete muting of the speech synthesizing in a receiver when data transfer connections containing a large number of transfer errors are used. This is due to the fact that when an error detection algorithm detects transfer errors in speech frames, it too easily mutes the speech synthesizer. This leads to the loss of speech information.
As is evident from the above description, there is a need to develop a better method of protecting information parameters on data transfer connections containing numerous transfer errors. In addition, there is a need to develop a system, the receiver of which better tolerates information parameter frames containing errors. In the following the information coding method according to the invention and the system utilizing it and the terminal devices are explained primarily using the speech coding in a mobile communication system as an example. Nothing however limits using the information coding system according to the invention for coding of data other than speech data. For the sake of clarity the invention is in the following also called a speech coding method, because it best describes one of the most important fields of application of the invention. It is possible to utilize the invention instead of a radio connection, also e.g. in connection with information transfer systems realized using wireline connections.
Now an information coding method utilizing focused error correction and error detection system has been invented, by use of which the above described problems can be reduced. One of the purposes of the present invention is to present a speech coding method which will be automatically adjusted as a function of the quality of a data transfer connection optimizing the speech quality on data transfer connections of any quality. The quality of the data transfer connection used is analyzed by measuring the parameters describing the quality of a data transfer connection, such as e.g. C/I (Carrier to Interference) ratio, S/N (Signal to Noise) ratio or bit error rate (Bit Error Rate, BER) as known to a person skilled in the art. In the information coding method according to the invention there is no need to reduce the number of bits used for speech coding in relation to the total bit rate used in the information transfer connection, in which case the voice quality of the speech preferably remains good. In the information coding method according to the invention error correction and/or error detection is focused on the bits most essential for voice quality as a function of the C/I-ratio or of some other parameter describing the quality of the data transfer connection. The muting of speech synthesizing occurring in prior art systems on data transfer connections of poor quality is reduced in the information coding method according to the invention by using the focused error detection, in other words using focused detection of bad frames.
The information coding method according to the invention tolerates data transfer errors well. The high tolerance of data transfer errors has been achieved by monitoring the quality of a data transfer connection, and by optimizing the focusing of the error correction and error detection of speech parameters. Both the error correction coding (e.g. convolution coding) and the error detection coding (e.g. checking of the cyclic redundancy) are adjusted to match the error conditions of the data transfer connection.
When few data transfer errors occur, all or almost all speech parameter bits are protected with an error correction code in a system according to the invention. When more data transfer errors occur, the error correction is focused more on the speech parameter bits most important for speech quality and intelligibility (when data other than speech is transferred, on the most important bits for the information). Alternatively, or additionally, when few data transfer errors occur, all or almost all speech parameter bits are protected with an error detection code in a system according to the invention, and when more data transfer errors occur, the error detection is focused more on the speech parameter bits most important for speech quality and intelligibility. The focusing of error detection bits can be done by keeping the number of error detection bits constant (e.g. 3 CRC bits) but performing error detection on a different amount of information bits depending on data transfer quality. The focusing of error detection bits can be done by varying the number of error detection bits depending on data transfer quality. Which bits that are the most important for speech quality is determined based upon the speech coding method used. For example, when a simple PCM (Pulse Code Modulation) coding is used, it is unambiguous that the most significant bits (MSB, Most Significant Bits) are more important and they must be protected carefully. The least significant bits (LSB, Least Significant Bits) can be left unprotected if needed, because their effect on the intelligibility of speech is small. In the FR-speech codec of the GSM-system, known to a person skilled in the art, the relative significance of the bits has been defined in the GSM-specification. The final decision of which bits are the most important for speech quality has been made subjectively, based upon listening tests. When in the method according to the invention, the error detection code is focused on the most important bits, the error correction code is at the same time changed to be more efficient, or more error correction information is included in relation to protected speech parameter bits. This is realized e.g. by using a convolution coding having a lower coding ratio.
When the quality of an information transfer connection becomes poor, as happens in a mobile communication system when the quality of a radio connection between a mobile station and a base station deteriorates, in a system according to the invention a smaller and smaller part of all speech parameter bits is protected selectively, but using an enhanced error correction code. The error correction coding is focused on the bits most important for speech quality in such a way that the most important bits are always protected and the less important bits are protected within the limits set by the quality of the information transfer connection and the number of bits available for error correction allow. As a result of the enhancement of the error correcting code, it is possible to decode speech in the receiving end even on information transfer connections containing numerous of errors, in other words, a system utilizing the speech coding method according to the invention will not xe2x80x9ccollapsexe2x80x9d, i.e., the output of the speech decoder is not muted. Bits transferred without the error correction code may reduce the speech quality due to transmission errors, but well protected, most important speech parameter bits still guarantee the intelligibility of the speech. As to speech quality, this method is far better than trying to protect all speech parameter bits or a major part of them using a poor error correction code. A poor error correction code on information transfer connections with plenty of interference results in a situation, in which the error correction code no longer is capable of correcting the transmission errors. In this case actually all bits used for error correction have been wasted. When the quality of the information transfer connection improves, or the number of transmission errors is reduced, the speech coding method according to the invention adapts itself correspondingly to the new situation and increases the share of speech parameter bits protected with the error correction code. Thus the error correction operates efficiently in all data transfer conditions.
It is possible to analyze the quality of an information transfer connection using a number of methods. Among the methods are the above mentioned ones known to a person skilled in the art, C/I (Channel to Interference)- and S/N (Signal to Noise)-ratios measured in an information transfer connection. It is possible to analyze the quality of an information transfer connection in a receiver also as a function of the frequency of occurrence of speech parameter frames rejected in the speech synthesizing due to errors contained in the most important bits, as it is explained below in more detail in connection with an embodiment of the invention. It is possible to perform the detection of the quality of an information transfer connection itself in both a transmitter and a receiver, but the information about the selected speech coding mode (or how the error correction and/or error detection bits are focused according to the invention) must always be transferred to the speech encoder of the transmitter.
A system according to the invention typically uses the same speech codec operating at a fixed line speed. Only the focusing of error correction coding and error detection coding is adapted to correspond with the current data transfer conditions. This facilitates the operation of the system above a xe2x80x9ccollapse pointxe2x80x9d. A collapse point means a situation in which a data transfer connection contains so many data transfer errors that a receiver no longer is capable of interpreting the received information. In other words, an information coding method utilizing the focused error correction and error detection according to the invention lowers the collapse point of a data transfer connection, which in practice means e.g. that a data transfer connection between a mobile station and a base station can be successfully established on radio connections with poorer Signal to Noise-ratios than previously.
A system utilizing the speech coding method according to the invention is automatically adapted according to the current data transfer conditions, and thus minimizes the effect of data transfer errors on the speech quality. Because the focusing of the error correction code on certain speech bits is a part of the error correction code itself, it is possible to realize, utilizing the invention, such a system operating at fixed line speed, which provides high speech quality. It is possible to use the invention on all kinds of data transfer connections without a need to go over to using a codec with a lower line speed. In such data transfer systems which use variable line speed or which change the ratio between speech parameter bits and error correction information, the invention can equally be utilized for improving the data transfer connection and through it the speech quality.
In addition to that in the speech coding method according to the invention the error correction code is focused in relation to the most important speech parameter bits, also the error detection code (for example CRC) is focused more to protect the most important speech parameter bits, the more data transfer errors the current data transfer connection contains. This further improves the probability of the information coming through and reduces the need to mute the speech signal in the receiver. This preferably improves the speech quality and intelligibility. When data transfer connections containing a very large number of data transfer errors are used, it is possible to accept data transfer errors in the less important bits of the speech frames and to use these speech frames for synthesizing speech in the receiver, because the quality of the speech is reduced anyhow if there has been a need to reject speech frames due to errors in the most important bits of the speech frames.
According to a first aspect of the invention there is provided a digital information transfer system, which comprises a transmitter and a receiver and an information transfer connection between said transmitter and receiver, said transmitter and receiver comprise means for the processing of information and for transferring it from said transmitter to said receiver using said information transfer connection, said transmitter comprises a channel encoder for processing the information, said information processing means comprise first dividing means for dividing the information into at least two parts, a first part and a second part, and that to said information of the first part in said channel encoder one of the following operations is performed: error correction coding for the correction of transfer errors occurring on the information transfer connection at reception, error detection coding for the detection of transfer errors occurring on the information transfer connection at reception, and a combination of them, the system being characterized in that it comprises means for analyzing the quality of said information transfer connection and for yielding a response describing the quality, and that said transmitter has been arranged to adjust the ratio between said first information and said second information based upon the response received from said information transfer connection quality analysis means.
According to a second aspect of the invention there is provided a mobile station comprising a transmitter and a receiver, said transmitter and receiver comprise means for processing the information, said transmitter comprises a channel encoder for processing the information, said information processing means comprise first dividing means for dividing the information into at least two parts, a first part and a second part, and that one of the following operations is performed in said channel encoder on said information of the first part: error correction coding for the correction at reception of transfer errors which occur in the information transfer connection, error detection coding for the correction at reception of transfer errors which occur in the information transfer connection, and a combination of them, the mobile station being characterized in that it comprises means for analyzing the quality of said information transfer connection and for yielding a response describing the quality, and that said transmitter has been arranged to adjust the ratio between said first information and said second information based upon the response received from said information transfer connection quality analyzing means.
According to a third aspect of the invention there is provided a transmitter comprising means for processing information and transferring it to an information transfer connection, said transmitter comprises a channel encoder for processing the information, said information processing means comprise a first dividing means for dividing the information into at least two parts, a first part and a second part and that one of the following operations is performed on said information of the first part in said channel encoder: error correction encoding for the correction at reception of transfer errors occurring in the information transfer connection, error detection encoding for the detection at reception of transfer errors occurring in the information transfer connection, and a combination of them, the transmitter being characterized in that it comprises means for analyzing the quality of said the information transfer connection and for yielding a response describing the quality, and that said transmitter has been arranged to adjust the ratio between said first information and said second information based upon the response received from said information transfer connection quality analyzing means.
According to a fourth aspect of the invention there is provided a receiver comprising means for receiving information from a transmitter over an information transfer connection, and a channel decoder for performing one of channel decoding operations on the information received from the information transfer connection, the receiver being characterized in that it comprises means for analyzing the quality of said information transfer connection and for forming a response based upon the performed analysis, means for determining the focused channel encoding mode to be used in said information transfer connection based upon said response, and means for transferring said channel encoding mode to the transmitter using said information transfer connection.
According to a fifth aspect of the invention there is provided a method for transferring information in a digital information transfer system, in which
the information to be transferred is divided for transmission into at least two parts, a first part and a second part,
on said information of the first part one of channel encoding operations is performed, the method being characterized in that
prior to the channel coding the quality of an information transfer connection is analyzed in order to form an analysis result, and
the ratio between said first part and second part is adjusted based upon said analysis result.